Tag: pbx in a flash

As an admin for a telephone system, possibly one of the most useful things you can do is monitoring your peers and trunks. There are 2 great reasons you should do so: 1. You can respond to and resolve issues with your system before your users know about it, and you can be in the know if someone reports “none of the phones are working” when in fact only 1 or 2 are not working 2. You can actually know when there is a problem with the system – where you otherwise might not know there is a problem until someone…

This information is provided without warranty – although I have been using this configuration successfully for over 12 months. In FreePBX there is a module which has changed it’s name but remains an extremely useful one. Day/Night control, now called Call Flow Control, allows you to set a toggle-switch to change how a call is routed within the system. Typically this could be a Day or Night mode service, but you might also want a ‘We are closed for Christmas’ message for example. Using an announcement as the ‘night’ destination, using a recording linked to a feature code before going…

The PBX in a Flash (PIAF) template provided when ordering a SysAdminMan VPS has been updated to version 2.0.6.2 Purple. This includes Asterisk 1.8.8.0 and FreePBX 2.8. Please see here for more information on a hosted PBX in a Flash virtual server – http://sysadminman.net/pbx-in-a-flash-hosting.html

When you start looking at control panels for Asterisk it can be difficult to decide what you should be using – FreePBX or A2Billing. While they are both web GUIs for setting up Asterisk, they are used for different things and which one to choose depends on your needs. Here is a brief description of both to help you decide – FreePBX Used for setting up extensions and trunks for inbound and outbound calls Includes lots of features of a traditional PBX – voicemail, IVRs, ring groups, queues etc. Includes Call Detail Records (CDR) that logs all calls, their destination…

If you’re using FreePBX or one of the distributions that use it such as Trixbox, Elastix, PBX-in-a-Flash and are having a problem with IVRs being slow to respond it it is worth checking that you do not have “Enable Direct Dial” enabled for the IVR. This option allows a customer to dial an extension number rather than an IVR menu option but this means that FreePBX has to wait to see if an extension number is being dialled, which can introduce a delay. If you don’t need callers to be able to dial extensions from an IVR then you can…

Conference calls are used a lot by businesses and are a built-in feature of Asterisk and FreePBX (and therefore the distributions that rely on these – Trixbox, Elastix, PBX-in-a-flash …) They are very easy to setup –

One of the things you need to do when looking for a server to run Asterisk on is figure out how much bandwidth you need for the number of concurrent calls you’re expecting to have. A great tool for this can be found here – http://www.asteriskguru.com/tools/bandwidth_calculator.php Just set the codec you’re going to be using (check with your VOIP provider – g.711/ulaw is usual and the highest quality), the connection type (usually SIP or IAX2 with Asterisk) and the number of concurrent calls. It will then display the bandwidth required for that many calls. One thing to watch out for…

EDIT – these test numbers are no longer functional As a quick demonstration of what you can achieve with Trixbox in a couple of hours I have put together a demonstration phone system. Trixbox uses Asterisk and FreePBX to provide a richly featured phone system that you can do lots of interesting things with. For the demonstration I created a phone system with DDI numbers in the London and New York. These phone numbers are provided by future-nine.com. If you would like do give it a go you can call the system using the following regular telephone numbers – UK …